It has unfortunately become a fact on this and other forums that once a thread seems to be going well and has a quite specific subject matter that someone will come in and tilt it off course. Usually this is accomplished by a challenge to the value of underlying and fundamental assumptions based on normally accepted laws, knowledge and practice.
To mention that one does not accept the basis of any statement is one thing, but to continue is another. At it's worst level it smacks of anti-social behaviour - at it's best it shows bad manners and a high level of misplaced self importance.
If one holds contrary views which have not been accepted as pertinent to the thread, then the correct behaviour is to start a new thread of one's own to discuss the matter with anyone who wishes to partake. Such continued disruptive intervention as we see here is not welcome to many. It seems as though for some that challenging those who give us good threads has become a substitute "blood sport".
EDIT: It seems that in this case that the following thread was started but got no-where as it was felt that the whole concept was not correctly presented:
http://www.diyaudio.com/forums/everything-else/173283-cmp-framing-what-you-mean.html
To mention that one does not accept the basis of any statement is one thing, but to continue is another. At it's worst level it smacks of anti-social behaviour - at it's best it shows bad manners and a high level of misplaced self importance.
If one holds contrary views which have not been accepted as pertinent to the thread, then the correct behaviour is to start a new thread of one's own to discuss the matter with anyone who wishes to partake. Such continued disruptive intervention as we see here is not welcome to many. It seems as though for some that challenging those who give us good threads has become a substitute "blood sport".
EDIT: It seems that in this case that the following thread was started but got no-where as it was felt that the whole concept was not correctly presented:
http://www.diyaudio.com/forums/everything-else/173283-cmp-framing-what-you-mean.html
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I certainly hope you have real data to elaborate such statement. I find that statement from Joachim quite realistic. You have to have some experience to find the optimum point for EQ.
There is no need for any addition data for rock solid law.
John Kreskovsky already has explained and outlined it numerous times around here and even in this thread - nothing to add.
Michael
It's easy to be doing the lip service while leaving the hard work to others
However, I think you have not seen anything from John ever since the Ultimate Equalizer became available.
When Joachim mentioned the possibility of equalization, I realized what I had overlooked. Really appreciate what bit of inspiration from him.
It's easy to be doing the lip service while leaving the hard work to others
However, I think you have not seen anything from John ever since the Ultimate Equalizer became available.
"Ultimate" or not - please do not forget that that UE from Bodzio is not the first one in the world that was available.
Short memory you seem to have...
May I remind you as an example regarding "lip service" and "ultimate" EQing (not claiming having been the first one, though):
http://www.diyaudio.com/forums/multi-way/100392-beyond-ariel-115.html#post1837992
An externally hosted image should be here but it was not working when we last tested it.
As they love to say : been there done that
Also - I'm pretty sure those measurements shown back then, triggered quite some general interest in the matter
BUT - since, I had a lot of time to think about those nice measurements - and possibly did invest more time to look *behind* that nice scene than anyone else...
Michael
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The more interesting thing would be what the original recording was. There are some recordings where different instruments seem to be recorded in different polarity. But this page you have posted a link to certainly seems to show that maybe the most accurate phase is a hit or miss situation.
I don't see the phase response."Ultimate" or not - please do not forget that that UE from Bodzan is not the first one in the world that was available.
May I remind you as an example regarding "lip service" and "ultimate" EQing :
http://www.diyaudio.com/forums/multi-way/100392-beyond-ariel-115.html#post1837992
An externally hosted image should be here but it was not working when we last tested it.
As they love to say : been there done that
Michael
So what did they do.An externally hosted image should be here but it was not working when we last tested it.
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George, regarding EQing - as said - you and most people are looking at the wrong places.
The interesting things happen with respect to spectral distribution being continuous or not (delay being involved or not).
*There* the wheat is spread from the cuff .
Anything else - as said :
Make FR as flat as can be and you are done. Definitely.
No need for any *experienced hand* regarding EQing - but a need for knowledge regarding whats actually going on
Michael
The interesting things happen with respect to spectral distribution being continuous or not (delay being involved or not).
*There* the wheat is spread from the cuff .
Anything else - as said :
Make FR as flat as can be and you are done. Definitely.
No need for any *experienced hand* regarding EQing - but a need for knowledge regarding whats actually going on
Michael
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Handwaving. If you have data accompanying listening difference description, then we can understand what you are saying. I agree with Joachim that reasonable experience is necessary, and I have already posted what I hear different. Of course, I am not showing everything, and do not plan to show such until I see other data related close to what you refer to.
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Please point to other software that does what the UE does. And I think, perhaps some history is in order. The ability to do what the UE does has been technology contained in SoundEasy for at lease 8 years. However, the process was some what involved and more laborious than desired. I have been using it over that time to explore linear phase system and "perfect equalization". Over the last year what has changed are several things. The UE was developed to simplify and automate the process. Then the UE was broken out of SoundEasy as a stand alone program with more highly developed interface, the ability to emulate up to a 4 way stereo pair of speakers, and more sophisticated DSP processing. Latency of the linear phase system is below 180 msec and for the amplitude only option, below 80 msec. These make the UE applicable to HT systems as well. The latency in linear phase mode is less that that of my Sony HT receiver.
As the current version of the UE stands, both amplitude and phase equalization are completely automatic. The user simply defines the desired acoustic target for each band pass, the frequency range over which the amplitude eq is to be applied, and then clicks a button to generate the digital filter. A quick look at how it works is presented here.
I've been using the UE for some time now and am working on a dipole system. I can attest to the simplicity. Prior to this I used the digital filter emulator in SE to design and audition crossovers prior to constructing them.
Here's a HOLM Acoustics measurement of that 3-way dipole. It's LR8 at 250Hz and 1200Hz. The room conditions don't allow for the W/M section to show. I set the crosshair at the Fc to examine it. As you can see, it's practically a perfect LR8 with the low leg deviation due to measurement conditions.
Note as well the phase of each leg. This was using the linear phase option. This is measured data, not modeled.
Dave
Yes, John's measurements don't show this, mine have, so I don't believe that it has to do with the UE. I'm using a DXT tweeter (non-dipole part of it) and have not found why I get this. It may have something to do with my measurement setup. This weekend I'll see if I can get my SoundEasy measurement going with this to compare.
Dave
No exactly hand waving – only different priorities regarding „what to learn“
You and me are just talking at different levels.
Some are more interested in „understanding“ with respect to the underlaying physics of sound creation.
So once more – making FR flat as flat can be and you are done.
This I said with respect to the B*S term „stored energy“ you focused on - and – with respect to the CSD plots you brought up „as prove“
This is pure physics - I certainly did *not* make any comment about psychoacoustics (perception)
*if* you want to go into that, there are lots and lots of threads eveywhere where people reinvent the wheel of Toole etc. on and on and on and on and on ….
There IMO is veeeery little to say to that topic :
from an individual standpoint its all very subjective and nothing, absolutely nothing, of all of this applies „like carved in stone“ - and if seen form a technical point of view, besides „FR“ there is only room and diffraction effects left which determine presentation – at least for non distorting sound radiation systems.
Seen form the ears point of view - LOL - room effects and diffraction effects are actually (to a vast degree) simple delay time effects – just at different scales.
FR + room effects + diffraction effects – all is embraced and merged in CMP concept – so what ?
What makes Joachims thread interesting for me is not what he is after in the end (nearfield monitoring to meet a certain experiance) – its the route he will take to arrive there and that he seems to have stumbled over the same delay issues - but - until now is rating them just as kind of „cepstrum analysis phenomena“.
So what I put into this thread – if any - is possibly a more focused way to identify trade offs to look at.
Hi John
I think the Behringer and several others do the same trick - CARA and some hack of CATT being the first convolution based SW I've used ( pretty old stuff
) - now I use PC based Accourate for the purpose of speaker response shaping and nowadays there are other PC based solutions of course - but - I'm not (was not and will not) arguing about any *details* of UE being capable of or not (going 6 way currently - its not any serieous option for me anyway). What I was referring to was just the general possibility of linearizing frequency and phase at will.If you wish, we can continue anytime where we got stuck - taking UE as the workhouse in exploring restrictions I claim with respect to correctability of CMP behaviour.
I will possibly throw in some additional wavelet analysis and Joachim – if got interested - may do some Cepstrum analysis too – so it might be juicy fun...
I'll have a look how to purchase UE standalone version and give it a try anyway (hope it works in my setup)...
Michael
PS
BTW, John your page linked is not correctly displayed on my Ubuntu/ Opera 11.01 (latest version) – old story ...
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From what I could tell the ringing in Dave's result is due to a resonance in the tweeter above the limit of UE equalization (metL dome). Here are some comparisons of a 2-way's response for both linear phase and the amplitude only eq. The linear phase speaker does very well at reproducing the reference impulse. Also inclused are off axis measurement to shoud the degradition on axis.
I have posted this before. Note that even off axis 30 degree the linear phase pulse is better than the standard LR4 pulse on axis.
An externally hosted image should be here but it was not working when we last tested it.
One of the things I don't feel comfortable with about the Holm software (is it HOLM impulse?) is that it does not show a reference impulse. In my setup, I recently just changed an interconnect in my setup to one tuned for flat impedance, the reference impulse at 96KHz sampling actually changed for the better.
I am more interested in seeing your data using whatever measurement and analysis tools you use for crossverification of what you talk about. Not handwaving statements like these. Sure, everyone is interested in learning, and everyone have their own background and experiences. John shows his, I show mine, Joachim shows his, etc. So why not let us see some of yours? If you have the anaysis plus measurement to show me wrong, fine. Please go ahead and show it.No exactly hand waving – only different priorities regarding „what to learn“
You and me are just talking at different levels.
Some are more interested in „understanding“ with respect to the underlaying physics of sound creation.
So once more – making FR flat as flat can be and you are done.
This I said with respect to the B*S term „stored energy“ you focused on - and – with respect to the CSD plots you brought up „as prove“
This is pure physics - I certainly did *not* make any comment about psychoacoustics (perception)
*if* you want to go into that, there are lots and lots of threads eveywhere where people reinvent the wheel of Toole etc. on and on and on and on and on ….
There IMO is veeeery little to say to that topic :
from an individual standpoint its all very subjective and nothing, absolutely nothing, of all of this applies „like carved in stone“ - and if seen form a technical point of view, besides „FR“ there is only room and diffraction effects left which determine presentation – at least for non distorting sound radiation systems.
Seen form the ears point of view - LOL - room effects and diffraction effects are actually (to a vast degree) simple delay time effects – just at different scales.
FR + room effects + diffraction effects – all is embraced and merged in CMP concept – so what ?
What makes Joachims thread interesting for me is not what he is after in the end (nearfield monitoring to meet a certain experiance) – its the route he will take to arrive there and that he seems to have stumbled over the same delay issues - but - until now is rating them just as kind of „cepstrum analysis phenomena“.
So what I put into this thread – if any - is possibly a more focused way to identify trade offs to look at.
Hi John
I think the Behringer and several others do the same trick - CARA and some hack of CATT being the first convolution based SW I've used ( pretty old stuff
If you wish, we can continue anytime where we got stuck - taking UE as the workhouse in exploring restrictions I claim with respect to correctability of CMP behaviour.
I will possibly throw in some additional wavelet analysis and Joachim – if got interested - may do some Cepstrum analysis too – so it might be juicy fun...
I'll have a look how to purchase UE standalone version and give it a try anyway (hope it works in my setup)...
Michael
PS
BTW, John your page linked is not correctly displayed on my Ubuntu/ Opera 11.01 (latest version) – old story ...
Personally, I always finally cross compare with changes in what I hear, whether my hearing is good or bad is a different issue. The point is if analysis predictions, measurements, and listening show a similar and clearly identifiable trend, to me, that is the right direction.
I was talking with a friend the other day about identifying changes in system performance. I used and analogy of noise in a quiet night; for the same noise during the daytime, you will not notice it, but in a quiet night, it may become very annoying. The same goes with stored energy, reduce that, and certain other distortion become more annoying. So the process is just to solve various imperfections little by little, and then one day you suddenly discover how much has been improved.
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Michael, i do not think it is sufficient to make the transfer function linear at one point in space. It whould be more ideal if this behaviour is maintained over a certain angle off axis so bundeling of the drivers has to be watched and optimum crossover frequencies have to be chosen too. Our experiment at Essex also has shown that you can get into trouble when you take a very steep digital crossover. Actually we found that a crossover steeper then L/R 4 gives rise to preringing off axis and destroys good off axis behaviour so i am not an advocate of extremely steep digital crossovers.
Yes John, i have seen too in dirs measuement that there is a tweeter resonance over 20kHz so we simply see the impulse response of that thruw the digital equaliser. Maybe a filter over 20kHz whould help
Yes John, i have seen too in dirs measuement that there is a tweeter resonance over 20kHz so we simply see the impulse response of that thruw the digital equaliser. Maybe a filter over 20kHz whould help
I guess the pre-ringing cant be avoided - depending on filter structure chosen
----
@ Joachim
The discussion about constant directivity is a different one - but of course is subject to your (and any other) speaker design - its implications have been discussed at leeeength around here.
BTW I'm not advocating high order slopes per se - I'm selecting my slopes as high as possibly by preserving 100% summation - which even with all that digital "little helpers" isn't any steep at all.
The order or steepness IMO is not the criterion here - again it depends on the (digital) filter structure chosen - simply put - the XO filters have to have perfect summation in the time domain and in the frequency domain (and should not go mad if summation isn't perfect - as this never is the case vertically off-axis) - that leaves only few choices from what Accourate offers.
Thats not necessarily a bad thing as brick wall filter - besides problems in the time domain - do not blend over very well IMO
The lowest slope limit IMO is the restrictions Doppler IM puts on a design
Michael
----
@ Joachim
The discussion about constant directivity is a different one - but of course is subject to your (and any other) speaker design - its implications have been discussed at leeeength around here.
BTW I'm not advocating high order slopes per se - I'm selecting my slopes as high as possibly by preserving 100% summation - which even with all that digital "little helpers" isn't any steep at all.
The order or steepness IMO is not the criterion here - again it depends on the (digital) filter structure chosen - simply put - the XO filters have to have perfect summation in the time domain and in the frequency domain (and should not go mad if summation isn't perfect - as this never is the case vertically off-axis) - that leaves only few choices from what Accourate offers.
Thats not necessarily a bad thing as brick wall filter - besides problems in the time domain - do not blend over very well IMO
The lowest slope limit IMO is the restrictions Doppler IM puts on a design
Michael
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